As the use of mobile or cellular phones becomes more widespread, the integration of phone capability is increasing rapidly, benefiting from the increase in the degree of semiconductor integration, typically exemplified by LSI (large-scale integration) devices.
The integration of several capabilities is apparent, for example, in a mobile phone capable of handling both telephone communications and music reproduction through a single phone apparatus, or even telephone, music playback, video recording, and audio recording capabilities.
The LSI device included as discrete components in the current mobile phone often can perform signal mixing of (a) voice signals, which are input on demand through a voice-sound input means, with reproduced music signals, or (b) other party's voice signals, which are demodulated after reception, with music signals. Following the mixing, resultant signals are sent either to a transmitting processing block or respective outputting means provided in the phone such as a speaker or a headphone.
FIG. 6 is a diagrammatic block diagram illustrating a known audio apparatus included in a mobile phone, which is adapted to perform voice signal processing (for example, Japanese Laid-Open Patent Application No. 2000-299718).
In the audio (inclusive of voice) apparatus of FIG. 6, digital voice signals SDVa, which are demodulated after reception, are converted by a voice D/A converter 101 into analog signals, subjected to volume control processing by an analog processing circuit 104, and output through a voice outputting means such as a first speaker SPa and others.
On the other hand, analog voice signals, which are input through a voice input means such as a microphone (MIC) and others, are subjected to volume control processing, additive (i.e., addition and subtraction) operation and other similar operation by an analog processing circuit 105, converted by a voice A/D converter 102 into digital signals SDVb, and sent to a transmitting modulation processing block.
The scheme of FIG. 6 is merely one of various examples for mixing voice and audio signals, and other alternative means maybe devised for the mixing process. The previous methods primarily utilize analog signal processing, which performs volume control processing and additive operation using operation amplifiers, in particular.
As a result, several difficulties can arise, such as degradation in analog signal characteristics caused by process fluctuation during operational-amplifier production, and occurrence of audible noises resulted from infusion of high frequency (noise) signals in the phone apparatus.
In order to obviate such difficulties, a method has been proposed, which is adapted to provide a D/A converter right before the outputting means such as a speaker and an A/D converter right after the voice inputting means such as a microphone, perform digital signal processing on both the digital signals prior to the conversion into analog signals for an output stage and on the digital signals following the conversion into digital signals by an input stage, and perform several operation and control such as additive operation and volume control processing.
As a result, since input signals such as digital voice signals and digital audio signals can be subjected to digital processing such as signal mixing and others directly in the digital domain without converting into analog signals, the degradation in signal characteristics can be prevented, which may otherwise occur when the digital signals are converted into analog signals.
FIG. 7 is a diagrammatic block diagram illustrating another known audio apparatus devised to attain such capability as mentioned just above.
Referring to FIG. 7, digital voice signals SDVa, which are generated by receiving and modulating digital voice signals previously received through sampling at a sampling frequency fs, are subjected by a first over sampling circuit Sc to an oversampling process at a frequency of a multiple of fs (for example, 4×fs).
Subsequently, resultant signals are sent to both a first digital processing circuit Da and a second sampling frequency conversion circuit Sb.
Following the over sampling process, the sampled digital voice signals SDVa are subjected in a first digital processing circuit Da to several processes such as volume control processing and desired signal bandwidth control processing, converted by a voice D/A converter Ca into analog signals SAVa, and output through a first speaker SPa as a voice outputting means.
In addition, the sampling frequency of the digital voice signals is converted by the second sampling frequency conversion circuit Sb, into the sampling frequency of the digital audio signals.
On the other hand, digital audio signals SDMa, which are regenerated through sampling at another sampling frequency Fs, are subjected by a second over sampling circuit Sd to an over sampling process at a frequency of a multiple of Fs (for example, 4×Fs). Subsequently, resultant signals are sent to both a third digital processing circuit Dc and a first sampling frequency conversion circuit Sa.
Following the oversampling process, the sampled digital voice signals SDMa are subjected in a third digital processing circuit Dc to several processes such as performing additive operation with digital voice signals, the frequency of which is converted by the second sampling frequency conversion circuit Sb, as mentioned just above, setting the ratio for the additive operation, adjusting sound volume, and controlling audio tone following the desired signal bandwidth control processing under predetermined program setting.
Thereafter, resultant signals are converted by an audio D/A converter Cc into analog audio signals SAMa, and output through a second speaker SPb and a headphone HP.
In addition, the sampling frequency of the digital audio signals is converted by the first sampling frequency conversion circuit Sa, into the sampling frequency of the digital voice signals.
Moreover, analog voice signals SAVb, which are input through a voice input means such as a microphone (MIC) and others, are converted by a voice A/D converter Cc into digital voice signals SDVb.
Following the conversion process, the converted digital voice signals SDVb are subjected in the second digital processing circuit Db to several processes such as performing additive operation with digital audio signals, setting the ratio for the additive operation, and adjusting sound volume.
Subsequently, the sampling of the resultant signals are performed by a downsampling circuit Se at a sampling frequency of one fourth of 4×fs, (i.e., fs) and sent to a circuit block (not shown) for performing transmission modulation.
In the frequency conversion with the above noted apparatuses, however, the degradation in audio characteristics has been encountered for the signals depending on their sampling frequency among plural kinds of signals, which are sampled for respectively at sampling frequencies different from each other.
More specifically, the degradation becomes more apparent in the case when the signals having a relatively large bandwidth are sampling-processed at frequencies of smaller bandwidth, since the Nyquist frequency having relatively narrow bandwidth limits the bandwidth of the original signals during digital filtering process. Because of the narrowed bandwidth, the audio characteristics of the signals attained by the following digital signal operation may be degraded compared with those obtained by analog signal processing.
It is therefore desirable to obviate the difficulty of the abovementioned degradation in audio characteristics caused by sampling processing at different sampling frequencies.